Does
anyone know an IPPhone or Device that works with
Asterisk as an announcement onlydevice with a loud speaker andthat
is online forever and will not hungup for any reason and even if it hangsup, it
will reboot itself to connect to a VOIP Server with a given configuration to
receive the
I
second that, except that if you have more money to spend and want a maintenance
free device, go for Cisco gear.
Qunitum needs a reboot once in a while. Cisco gear does
not. There is some difference in call quality and call handling as
well.
Seshu
Kanuri
From: [EMAIL PROTECTED]
I totally agree. But doe the Asterisk list servers have any such feature
to block the spam and delete the spamming users? I don't think so.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Tuesday, November 08, 2005 2:13 PM
Nrk,
Do some googling and try to find all this info on the Wiki site for
Asterisk.
SK
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nr k
Sent: Sunday, November 06, 2005 5:20 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] how to
The
Thirdlane PBX Manager solution is just a few perl scripts. This is no better
than what you can do by directly modifying the Asterisk Config files or many
Open Source GUIs like Phonecall etc you have out there.
Infact
Areski's A2Billing has a good extension configurator in the
You
can also try http://www.terracall.com I have been using them with good results
lately
Seshu
Kanuri
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
BrashearSent: Thursday, November 03, 2005 11:03 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users]
I want to give the benefit of doubt to the suggestion as I think there
is a misunderstanding of the suggested method of removal of AMP.
I guess that he was suggesting to remove it from your linux installation
by using the rm -rf command as under
cd /var/www
rm -rf *
which will efectively
interested in knowing what it takes for Asterisk to be
able to handle the 100 channels I need to run
Simultaneously.
Seshu Kanuri
-Original Message-
From: Iain Barker [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 02, 2005 1:41 PM
To: Kanuri, Seshu (Company IT)
Subject: Re
Add the line as under to your sip.conf entry.
Progressinband=no
;progressinband=never - is erroneous
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Miller
Sent: Tuesday, November 01, 2005 2:23 PM
To: Asterisk Users Mailing List -
Folks!
I want to know if anyone in the list is using OpenSER,
which appears to be a fork of SER. If so can you post
Your comments on its functionality?
The location where this is available is here:
http://openser.org/index.php#about
Some of the the features I am impressed with being...
Areski,
The featurelist in this version is Awsome. This will clearly and
absolutely make all those Closed Source Billing systems and so called
Soft Switches like Bicom's Switchware, obsolete.
Kudos for your effort and contribution to the Asterisk users.
This probaly is one of the most
[EMAIL PROTECTED] wrote
It's a free service. It belongs on this list.
Olle is right. Even if it is a free service it does not belong here.
This forum is for any Asterisk related user issues, not some DID issue
of one of a hundred such service providers.
Take it off this list.
Seshu Kanuri
1)What is the protocol you are using? SIP or IAX2?
2)Have you applied the correct firmware to the Phone?
Pa168 phones are falwless when connecting to Asterisk.
Start the configuration as asimple entry as under.
I have added Port address and allowed codecs in the config below:
[221]
with this phone, but my hope was that with the
new firmware something could be solved.
Thanks again.
2005/10/13, Kanuri, Seshu (Company IT) [EMAIL PROTECTED]:
1)What is the protocol you are using? SIP or IAX2?
2)Have you applied the correct firmware to the Phone?
Pa168 phones are falwless
Are has answered this question with an example, which shall statisfy the
curious.
-Original Message-
Kanuri, Seshu (Company IT) wrote:
This is relevant where Administrative users wanted to manage their
Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc
Now you have made me
.. there is no fancy things in rating
like tariffs or similar... (at this moment)
You can check on www.slsolucije.hr .. it is on Croatian.. but..
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT)
Sent: Monday, October 10, 2005
There was some discussion in the past about which one is the best
Content Management System that can be used in conjunction with Asterisk.
Mambo was supposed to be the best out there under GPL. The guys who
developed Mambo have a new product now - Joomla. I am using this and it
appears to be
And how exactly is Asterisk relevant to a CMS? could you give a more
specific example?
This is relevant where Administrative users wanted to manage their
Asterisk GUI setups like [EMAIL PROTECTED], AMP, Phonecall etc
Seshu
NOTICE: If
Looks like Terracall
has not only reduced the rates but also reduced their ability to connect the
calls to India.
Today we are not
able to make even one call, but the CDRs are still coming as connected and we
are being charged.
Please note the
request I sent below for the credit.
Could not agree more with Matt. I have been a linux geek for a long time
and I would think twice before calling Windows a crap o/s as linux feels
crappier when it comes to usability, administration and the pain in
making it work the first time, with due respect to all those who are
contributing to
: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT)
Sent: 27 September 2005 19:07 To: Asterisk Users Mailing List -
Non-Commercial Discussion Subject: RE: [Asterisk-Users] Software only
Asterisk PBX (commercial) Don't you ever recommend Bicom
Morgan,
For the most part people have either ignored or dismissed this company
Bicom Systems and it's products Pbxware and Switchware as they don't fit
into the same mould as Asterisk / Linux / APACHE/ Mysql /PHP /SugarCRM.
Bicom Systems want to ride on the Open source systems without giving
back
The religious Zealot was catholic or more accurately speaking, a
Zehova's witness
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of canuck15
So how does that explain muslims blowing themselves up and taking as
many non-believers with them as possible?
Alberto,
PA168
chip does not have Hold and Transfer features on it until firmware version 1.44.
Atcom never claimed that
these
will work as the Pa168 firmware is still under development.
Yesterday I met Peter Sun, President and owner of Atcom
China, in New York. He is here toattend VON
Don't you ever recommend Bicom as they take your money and will never
deliver a product that works.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Morgan
Gilroy
Sent: Tuesday, September 27, 2005 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial
Very rarely we come across real success stories using Asterisk as a part
of a great solution, and when I see one, I want to share it with you.
Though it is not mentioned in the news item, it is a fact that Iareanet
uses Asterisk as the core for their messaging part of the solution and
today they
USB phone and NAT - What has USB Phpne got to do with NAT?
USB Phone is just a hardware piece that pipes the audio output from your
softphone.
Your softphone has to take care of that.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Callum
What can RAGI do additionally that AGI or FastAgi and DeadAgi cannot do
which is already available under Asterisk?
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of joe
heitzeberg
Sent: Sunday, September 11, 2005 12:31 PM
To: Asterisk Users
As an ex 7th employee of PayPal before it was sold to ebay and having
gone through Ebay's carenot attitude to it's customers, let alone
investors, I concur with Dean's comments on ebay's future.
If you still have ebay's shares in your closet, get them out now.
Seshu Kanuri
Dean Collins Wrote:
That theory sounds great on paper, but consider some of the factors
below.
1) On the web there are no loyalties. Today it is Skype but tomorrow
certainly belongs to GoogleTalk, if they can put Encryption. How
difficult is that to do? Whose quality is better? Skype or Googletalk? I
read that
Ebay is seething with rage as yahoo took 40% stake in Alibaba in China.
They want to counter that by acquring a large user base in Asia, so that
their experience in Japan (Where ebay was kicked out completely by local
players) is not repeated in China and India.
This is an example of how you
Skype will utilise PayPal for all payment systems.
What prevents Skype users using PayPal even without Ebay?
Ebay doesn't have lots of users in territories that Skype does (Nordic
and South America for one).
Ebay is not interested in those territories anyway as they don't operate
Paypal in
The article below, posted a while ago by a Wharton dude is very predictive and
interesting on Skype's power.
http://www2.cio.com/higher/report3799.html
Some snippets from the article are pasted here:
--
Skype's potential has a few big investors
Good job Coalescent Systems. Good Job Jason Becker and Ryan Courtnage.
You guys rock.
Your open source AMP GUI is more feature rich and much more desirable
than some closed source garbage out there in the market.
Keep up the goodwork.
Seshu Kanuri
-Original Message-
From: [EMAIL
Have a
look at the article appearing in Globalsources, pastedbelow, that
highlights the growing marketshare for IP PBX.
http://www.globalsources.com/gsol/GeneralManager?design=cleanlanguage=enpage=showarticleaction="">
Seshu
Kanuri
NOTICE: If received in error, please destroy and
And to add to what Kevin said, we don't want any closed source stuff, be
it a database module or a device driver, to be a part of Asterisk as a
standard module, for obvious reasons.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
I use BINK to burn ISO Images and it works great.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: Tuesday, August 30, 2005 11:09 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] (no subject)
On Tue,
I will not use any VOIP service that requires a large
upfront payment, in this case a 1 year service charge
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mag
GamSent: Wednesday, August 31, 2005 10:57 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
Atcom's ATA AG168 can do this. Contact the US distributor
http://www.iareaphone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Henderson
Sent: Monday, August 22, 2005 8:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Isnt Firefly and for that matter any other IAX2 Softphone an IAX2
Endpoint in real sense?
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Walsh
Sent: Saturday, August 20, 2005 7:08 AM
To: Asterisk Users Mailing List - Non-Commercial
Dustin,
It is pretty amazing, that you PhoneCALL has so many features
incorporated into a GUI of the tool, that needs little manual
modifications to the Asterisk config files.
I am sure that this will make all those closed source Commercial GUIs
redundant in near future.
Kudos and keep up the
Sherwood,
Your intentions are noble and your desire to build this, fullfills an
immediate need for business.
If your intention is just to build a GUI for Asterisk, read no further.
If your desire is to build something more purposeful, your best bet
would be to see the existing commercial
Senad,
I don't want to take this conversation much further and send a laundry
list of issues we faced with Switchware to this forum, and the myriads
of bugs we have wrestled with in the past 18 months, as that list is too
long and this forum is not for that purpose.
In addition, our clients we
As this thread has come into the open, my suggestion is to get at least
5 references for Swithware and 5 references for Pbxware from Bicom
Systems, and speak to all of them and decide which way to go.
I can probably give a couple of references you can speak to, besides
myself on the usability of
Michael,
Here are some of the reactions to your original post on the T38 FAX
thingy:
1) Why the big secret? Why not post your solution to the list?
2) It's probably just another one of those nasty closed source add-ons
for sale.
3) I'm guessing it has nothing to do with *. Probably MAX TNT,
You may have bought the Chinese Versions and hence the problem in slow
response.
Have you tried the US versions available from http://www.iareaphone.com
?
-S
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Redstone
Sent: Tuesday, August 02, 2005
Graham,
Digium IAX2 FXS unit called IAXY is just no good. I would say that it is
garbage.
Try the IAX2 ATA ( AG168 sold as Netweb ATA-100) with a life line port
made by Atcom and available from http://www.iareaphone.com
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
With due respect to Digium and Mark Spencer and the greatest protocol he
defined, I have used IAXY and I regret to say that IAXY at $99 is plain
garbage compared to the $49 ATA made by ATCOM.
Try the ATCOM AG168 sold as ATA-100 by iareaphone.com. This has an
additional lifeline port and gives
Use Googleextensively andthe WIKIsitehere http://www.voip-info.org/wiki-Asterisk,
till you become familiar with the architecture of Asterisk. probably for a
couple of months.
You can come back here if you still have any questions
at that time and all the member here would be happy to
I am using asterisk-oh323-0.7.2-pre and CVS Head of Asterisk.
Oh323 Module compiled without errors. But When I try to stary Asterisk
with the Oh323.so file in the modules folder, Asterisk is dying with the
following error.
[chan_oh323.so]Aug 2 14:08:14 NOTICE[18873]: res_musiconhold.c:490
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SIP WEB Phone (Wanna implement Click to
Call)
On Thu, 2005-07-28 at 10:48 -0400, Kanuri, Seshu (Company IT) wrote:
Try babar nazmi's IAX web phone. This does not have G729 or G723 but
it has high bit rate
Try babar nazmi's IAX web phone. This does not have
G729 or G723 but it has high bitrate codecs.
http://www.geocities.com/babarnazmi/
Weatiareanetusethisproductaspartofourvirtualofficesolutionwhereremotecustomersdialin and dial out
usingtheIAXSoftPhone.
Seshu
From: [EMAIL PROTECTED]
Folks!
I would appreciate if someone could send me a simple working h323
configuration file oh323.conf that is part of [EMAIL PROTECTED]
installation.
I have tried to use the oh323.conf content listed on WIKI but it is just
not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot
Title: Soft Phone
Firefly Third Party version beats all others.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
MorrowSent: Friday, July 22, 2005 4:12 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] Soft Phone
Can anyone
I agree with Brian! Robert's post is off topic or
may be just a marketing effort, to push their site.
Anyone who wants freshtel.net for US/Canada calling
at 6.9 Cents a minute, raise their hands?
...
I see none
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
It does not look like Nufone is still in business, judging from the
content on their site, which is very little. There is not even a
configuration document to download, to connect to their network.
The rates file is only for US/Canada calling. No international
rates on this rates.csv file.
I
What is the Model you have? ML220(newer model, Supports SIP and H323) or
ML210A (older model, supports only H323 I guess)?
The manual at the link below is self explanatory. Just provide your Server IP,
Account and Pin/Password for your Asterisk Box and you are ready to go.
Seshu
What happens if you never received the first email from Nufone when you
signup? Is there someway to get this information from the web site? I
don't even see a download area for such information.
Can someone please send me nufone server address off list?
Seshu
-Original Message-
From:
1) reinvite=yes is incorrect syntax? Check the info here:
http://voip-info.org/wiki-Asterisk+sip+canreinvite
You can keep canrenvite=yes, but why do you want that?
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go
TNT is the worst piece of garbage that has ever been sold in the name of
a VOIP Switch.
This stuff is avialble on ebay for a fraction of it's sticker price, if
you dare to bid on it.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Try www.SIPphone.com or www.terracall.com
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ellafi
FituriSent: Tuesday, July 05, 2005 2:07 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users] SIP PHONE
Hi All,
I just got
The price of $39 for a WIFI SIP Phone sounds goofy to me.
I am not sure if you will be able to buy anything that is
close to a working wired SIP Phone, let alone a WIFI one.
I have a couple of Zyxel phones and they cost 5 times this price.
It looks like the makers of Hop1502 are trying to
How About Pro Reliable before we look for Anti Explosive WIFI
phones.
Has anyone got recommendations on WIFI Phones that work with Asterisk?
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Wednesday, June 15, 2005 4:53 PM
To:
[EMAIL PROTECTED] will
not be able to configure polycom500 phones.
You need to add this entry in sip.conf manually with
one additional line as under:
progressinband=no
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
MillerSent: Monday, June 13, 2005 9:44
Mark,
This is a wonderful thing to do for underserved societies like Uganda.
The datasheet you have provided and the layout could be the model for
many other developing societies both In Africa as well as central and
South America.
Kudos to Inveneo.org under your able leadership. Keep up the
Folks!
I have this
expensive gizmo Zyxel-2000 WIFIWireless Phone that can run SIP
protocol.
I have configured
this to my Asterisk as a SIP client but cannot register at the
server.
I have a basic
configuration entry in sip.conf and I am running it having the client
connected
with a
Voxee will not accept any calls that are not in G729.
You need G729 codec on your Asterisk.
Period.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, June 09, 2005 10:08 AM
To: Asterisk Users Mailing List -
.
On Mon, 6 Jun 2005 16:41:43 -0400, Kanuri, Seshu (Company IT) wrote
Kristian,
I am talking about your distro, that does not seem to be able to boot
when I have mounted (if that is the right word) the CF into my Dell
Server and tried to boot from it as the only IDE drive available
This Phone is same as Netweb-X100 made by Yuxin. These phones are
reliable. It has PA168 Chipset.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Brown
Sent: Wednesday, June 08, 2005 5:28 PM
To: Asterisk Users Mailing List - Non-Commercial
Your ringtone seem to have gone bad. You have to upload a new ringtone
file to correct your phone problem.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of snacktime
Sent: Wednesday, June 08, 2005 5:21 PM
To: Asterisk Users Mailing List - Non-Commercial
Folks!
If you are using Polycom Phones, any model - 500, 300, 400, 600 etc,
please rememeber to add this line to your sip.conf entry.
Progressinband=no
Without this line, these phones may not work. Probably this one line may
fix most of the problems users are reporting on this forum about
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kanuri,
Seshu (Company IT)
Sent: Monday, June 06, 2005 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] How to make Polycom phones work with Asterisk
as aSIP Client?
Folks!
If you are using Polycom Phones
Abel,
In have the same issue when I have burned the image to an 800MB CF Disk.
All it displays is GRUB CLI in a continuous stream.
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of abel
Sent: Monday, June 06, 2005 2:22 PM
To: Asterisk Users Mailing
I have never used inband. I always used
rfc2833.
This problem is seen even if you are using
dtmf=rfc2833.
the line progressinband=no, fixes
that.
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
MillerSent: Monday, June 06, 2005 3:07 PMTo:
Kristian,
I am talking about your distro, that does not seem to be able to boot
when I have mounted (if that is the right word) the CF into my Dell
Server and tried to boot from it as the only IDE drive available.
The Linux just does not kick in.
If you want to debug this I can Fedex to you,
Try ATCOM's AG168V available from US
Distributorhttp://www.iareaphone.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
DineshSent: Wednesday, June 01, 2005 9:40 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] IAX2 analog
telephone adapter
Hello
All,
Remove the Tthr options. You don't need any of them in the dial string for
AT320s
Seshu
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis
Sent: Wednesday, June 01, 2005 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I don't see the SugarCRM being part of the install.
How do you activate this?
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Tuesday, May 31, 2005 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Robson,
This is a Frames probelm. Areskicc uses a frame on the
left and this frame may be showing your Login page in it again, instead of the
menu. If this is the case, please check the "Stupids Guide to AreskiCC" on the
WIKI
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Chris,
This is a good beginning. Though this does sound useful to me, I would
also like to mention that there are some database management tools
already available out there - phpMyAdmin for example, where you can
modify everything visually and insert/update/delete Asterisk table data
over the
]
Sent: Wednesday, May 25, 2005 1:03 PM
To: Kanuri, Seshu (Company IT)
Subject: RE: oh323 problems
Thanks a million Seshu, it worked like a champ. Thanks
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
To: Tola Ogunsan [EMAIL PROTECTED]
Subject: RE: oh323 problems
Date: Tue, 24 May 2005 10:11:16
Let me recollect what I needed:
1) We need a TRUNK Configurator that can easily create multiple SIP/IAX
trunks and assign them to the dialing contexts. Current GUI tolls are
not doing this properly.
2) We need DUNDI configurator for Inter-Server access
3) We need an Accounting Module that can
Mike,
Many of the providers I've tried contacting either
won't call me back, or want me to sign an NDA just
to get a rate quote, or some other bullshit.
Assuming that you will need about 12 to 24 simulataneous calls on each
DID you want to run, and you are using Ulaw to get these calls,
Title: Message
FireFly is the best of the IAX softphones. Other
softphonesdo not work as good as FireFly. DIAX has many bugs still. DIAX
Softphone disconnects with Windows DLL errors everytime there is a problem in
the call like Asterisk Channel Not available etc.
Seshu
From: [EMAIL
Try changing SetCIDNum SetCallerID and use to SetCIDName as under:
Ex:
---
exten = s, 1, SetCallerID(${CALLERIDNUM})
exten = s, 2, SetCIDName(${CALLERIDNAME})
exten = s, 3, Dial(${ARG2}/${ARG1},${RINGSECS})
exten = s, 4, Voicemail(u${ARG1})
exten = s, 5, Hangup
exten = s, 101, Voicemail(b${ARG1})
Forget the dialogic, the drivers are old and not free and almost no-
one is using them.
I concur that view. Don't even touch Dialogic garbage. I have see people
spending months without being able to make them work. Dump them into
your incinerator and turn the knob from 'Medium' to 'High'.
I want to add to this question as this seems relevant here. My question
corresponds to sip.conf, iax.conf and [globals] in extensions.conf. My
questions are as under:
1) Can we have two general sections - one in sip.conf and another one in
one of the included additional sip extension files, like
Frank,
Your solution is not clear to me. Can you tell me what Step 2 will do?
[general]
register = [EMAIL PROTECTED]
How will it resolve the name obelix as an authenticated user, assuming
that asterix is reolved using dns?
Seshu
-Original Message-
From: [EMAIL PROTECTED]
Folks!
I am looking at a couple of models of Fixed GSM Gateways for the Purpose of
VOIP connectivity and specifically to work with Asterisk. I found that these
can be imported into USA for about $99.99 or about that. This is a one channel
unit just like tellular, one of them has GPRS.
Interactive Intelligence has a commercial Speech recognition API for
this purpose.
Check http://www.inin.com
Or the specific Vocalite engine page at:
http://www.inin.com/Products/vocalite/vocalite.asp
Seshu Kanuri
NOTICE: If received
Title: * Server
http://www.iareaphone.com sells
these
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Montague,
ClarenceSent: Thursday, May 12, 2005 1:30 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] *
Server
Any reviews/comments out there on this
What do you mean Requires PHP+pear+php/mysql. But Will run as CGI. I
have had it working with php. So apache is not required.
To make PHP work, Apache is required anyway as a web server. Is in't it?
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
I have followed the
Idiots' guide for installation, but still could not make it
work.
When I try to login
at the web page coming from /var/www/html/areski , I get the following
errors:
Can some body give me some hints where and
what to check for this error?. I am looking for info on the
Nabeel,
I am trying to install AreskiCC
and I get the following errors.
Warning: pg_pconnect():
Unable to connect to PostgreSQL server: could not connect to server: Connection
refused Is the server running on host localhost and accepting TCP/IP connections
on port 5432? . in
Hi Tim,
I am interested in your half finished Java AIX2 App. Can you send me the
source code?
Seshu Kanuri
212-537-2849
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of tim panton
Sent: Wednesday, May 11, 2005 4:28 PM
To: Asterisk Users Mailing List -
Vikram,
Instead of trying to be over-ambitious and try to connect 20 Asterisk
boxes together, why don't you try top connect three (3) of them together
first.
There may lie a plausible solution for you. If this is done, you may go
and string four of them together and so on and so forth.
Take the
Alex,
Asterisk does not have aOutbound SIP Proxy.
Remove any Proxy configuration from your Phone. I guess that part iscalled
Registrar Server.
Omit that information here and it should
work.
Seshu
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Folks!
Let me clarify this for you all. ATCOM's ATA does not have an FXO port.
The Lifeline port is not an FXO Port. It is an FXS Passthrough port.
It does not have any of the FXO features that you are looking for. You
cannot do a modprobe on this - nor can you pass your peer traffic to
this
[EMAIL PROTECTED] Wrote:
--
-Original Message-
Check the IareaNet solution at http://www.iaraenet.net
I hope the IareaNet solution works better than the above link.
Paul,
Sorry for the typo. Try http://www.iareanet.net
Seshu
Henry Devito wrote:
---
I've since migrated away from [EMAIL PROTECTED], but the codes are there in the
extensions.conf file.
I am also cusrious to know as to what you have migrated to, from
[EMAIL PROTECTED]
Seshu
-Original Message-
From: [EMAIL PROTECTED]
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